dpcm.c 12 KB

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  1. /*
  2. * Assorted DPCM codecs
  3. * Copyright (c) 2003 The ffmpeg Project
  4. *
  5. * This file is part of FFmpeg.
  6. *
  7. * FFmpeg is free software; you can redistribute it and/or
  8. * modify it under the terms of the GNU Lesser General Public
  9. * License as published by the Free Software Foundation; either
  10. * version 2.1 of the License, or (at your option) any later version.
  11. *
  12. * FFmpeg is distributed in the hope that it will be useful,
  13. * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14. * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
  15. * Lesser General Public License for more details.
  16. *
  17. * You should have received a copy of the GNU Lesser General Public
  18. * License along with FFmpeg; if not, write to the Free Software
  19. * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20. */
  21. /**
  22. * @file
  23. * Assorted DPCM (differential pulse code modulation) audio codecs
  24. * by Mike Melanson (melanson@pcisys.net)
  25. * Xan DPCM decoder by Mario Brito (mbrito@student.dei.uc.pt)
  26. * for more information on the specific data formats, visit:
  27. * http://www.pcisys.net/~melanson/codecs/simpleaudio.html
  28. * SOL DPCMs implemented by Konstantin Shishkov
  29. *
  30. * Note about using the Xan DPCM decoder: Xan DPCM is used in AVI files
  31. * found in the Wing Commander IV computer game. These AVI files contain
  32. * WAVEFORMAT headers which report the audio format as 0x01: raw PCM.
  33. * Clearly incorrect. To detect Xan DPCM, you will probably have to
  34. * special-case your AVI demuxer to use Xan DPCM if the file uses 'Xxan'
  35. * (Xan video) for its video codec. Alternately, such AVI files also contain
  36. * the fourcc 'Axan' in the 'auds' chunk of the AVI header.
  37. */
  38. #include "libavutil/intreadwrite.h"
  39. #include "avcodec.h"
  40. #include "bytestream.h"
  41. typedef struct DPCMContext {
  42. int channels;
  43. int16_t roq_square_array[256];
  44. int sample[2]; ///< previous sample (for SOL_DPCM)
  45. const int8_t *sol_table; ///< delta table for SOL_DPCM
  46. } DPCMContext;
  47. static const int16_t interplay_delta_table[] = {
  48. 0, 1, 2, 3, 4, 5, 6, 7,
  49. 8, 9, 10, 11, 12, 13, 14, 15,
  50. 16, 17, 18, 19, 20, 21, 22, 23,
  51. 24, 25, 26, 27, 28, 29, 30, 31,
  52. 32, 33, 34, 35, 36, 37, 38, 39,
  53. 40, 41, 42, 43, 47, 51, 56, 61,
  54. 66, 72, 79, 86, 94, 102, 112, 122,
  55. 133, 145, 158, 173, 189, 206, 225, 245,
  56. 267, 292, 318, 348, 379, 414, 452, 493,
  57. 538, 587, 640, 699, 763, 832, 908, 991,
  58. 1081, 1180, 1288, 1405, 1534, 1673, 1826, 1993,
  59. 2175, 2373, 2590, 2826, 3084, 3365, 3672, 4008,
  60. 4373, 4772, 5208, 5683, 6202, 6767, 7385, 8059,
  61. 8794, 9597, 10472, 11428, 12471, 13609, 14851, 16206,
  62. 17685, 19298, 21060, 22981, 25078, 27367, 29864, 32589,
  63. -29973, -26728, -23186, -19322, -15105, -10503, -5481, -1,
  64. 1, 1, 5481, 10503, 15105, 19322, 23186, 26728,
  65. 29973, -32589, -29864, -27367, -25078, -22981, -21060, -19298,
  66. -17685, -16206, -14851, -13609, -12471, -11428, -10472, -9597,
  67. -8794, -8059, -7385, -6767, -6202, -5683, -5208, -4772,
  68. -4373, -4008, -3672, -3365, -3084, -2826, -2590, -2373,
  69. -2175, -1993, -1826, -1673, -1534, -1405, -1288, -1180,
  70. -1081, -991, -908, -832, -763, -699, -640, -587,
  71. -538, -493, -452, -414, -379, -348, -318, -292,
  72. -267, -245, -225, -206, -189, -173, -158, -145,
  73. -133, -122, -112, -102, -94, -86, -79, -72,
  74. -66, -61, -56, -51, -47, -43, -42, -41,
  75. -40, -39, -38, -37, -36, -35, -34, -33,
  76. -32, -31, -30, -29, -28, -27, -26, -25,
  77. -24, -23, -22, -21, -20, -19, -18, -17,
  78. -16, -15, -14, -13, -12, -11, -10, -9,
  79. -8, -7, -6, -5, -4, -3, -2, -1
  80. };
  81. static const int8_t sol_table_old[16] = {
  82. 0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
  83. -0x15, -0xF, -0xA, -0x6, -0x3, -0x2, -0x1, 0x0
  84. };
  85. static const int8_t sol_table_new[16] = {
  86. 0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
  87. 0x0, -0x1, -0x2, -0x3, -0x6, -0xA, -0xF, -0x15
  88. };
  89. static const int16_t sol_table_16[128] = {
  90. 0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,
  91. 0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,
  92. 0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,
  93. 0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,
  94. 0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,
  95. 0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,
  96. 0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,
  97. 0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,
  98. 0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,
  99. 0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,
  100. 0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,
  101. 0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,
  102. 0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000
  103. };
  104. static av_cold int dpcm_decode_init(AVCodecContext *avctx)
  105. {
  106. DPCMContext *s = avctx->priv_data;
  107. int i;
  108. if (avctx->channels < 1 || avctx->channels > 2) {
  109. av_log(avctx, AV_LOG_INFO, "invalid number of channels\n");
  110. return AVERROR(EINVAL);
  111. }
  112. s->channels = avctx->channels;
  113. s->sample[0] = s->sample[1] = 0;
  114. switch(avctx->codec->id) {
  115. case CODEC_ID_ROQ_DPCM:
  116. /* initialize square table */
  117. for (i = 0; i < 128; i++) {
  118. int16_t square = i * i;
  119. s->roq_square_array[i ] = square;
  120. s->roq_square_array[i + 128] = -square;
  121. }
  122. break;
  123. case CODEC_ID_SOL_DPCM:
  124. switch(avctx->codec_tag){
  125. case 1:
  126. s->sol_table = sol_table_old;
  127. s->sample[0] = s->sample[1] = 0x80;
  128. break;
  129. case 2:
  130. s->sol_table = sol_table_new;
  131. s->sample[0] = s->sample[1] = 0x80;
  132. break;
  133. case 3:
  134. break;
  135. default:
  136. av_log(avctx, AV_LOG_ERROR, "Unknown SOL subcodec\n");
  137. return -1;
  138. }
  139. break;
  140. default:
  141. break;
  142. }
  143. if (avctx->codec->id == CODEC_ID_SOL_DPCM && avctx->codec_tag != 3)
  144. avctx->sample_fmt = AV_SAMPLE_FMT_U8;
  145. else
  146. avctx->sample_fmt = AV_SAMPLE_FMT_S16;
  147. return 0;
  148. }
  149. static int dpcm_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
  150. AVPacket *avpkt)
  151. {
  152. const uint8_t *buf = avpkt->data;
  153. int buf_size = avpkt->size;
  154. const uint8_t *buf_end = buf + buf_size;
  155. DPCMContext *s = avctx->priv_data;
  156. int out = 0;
  157. int predictor[2];
  158. int ch = 0;
  159. int stereo = s->channels - 1;
  160. int16_t *output_samples = data;
  161. if (!buf_size)
  162. return 0;
  163. /* calculate output size */
  164. switch(avctx->codec->id) {
  165. case CODEC_ID_ROQ_DPCM:
  166. out = buf_size - 8;
  167. break;
  168. case CODEC_ID_INTERPLAY_DPCM:
  169. out = buf_size - 6 - s->channels;
  170. break;
  171. case CODEC_ID_XAN_DPCM:
  172. out = buf_size - 2 * s->channels;
  173. break;
  174. case CODEC_ID_SOL_DPCM:
  175. if (avctx->codec_tag != 3)
  176. out = buf_size * 2;
  177. else
  178. out = buf_size;
  179. break;
  180. }
  181. out *= av_get_bytes_per_sample(avctx->sample_fmt);
  182. if (out < 0) {
  183. av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
  184. return AVERROR(EINVAL);
  185. }
  186. if (*data_size < out) {
  187. av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
  188. return AVERROR(EINVAL);
  189. }
  190. switch(avctx->codec->id) {
  191. case CODEC_ID_ROQ_DPCM:
  192. buf += 6;
  193. if (stereo) {
  194. predictor[1] = (int16_t)(bytestream_get_byte(&buf) << 8);
  195. predictor[0] = (int16_t)(bytestream_get_byte(&buf) << 8);
  196. } else {
  197. predictor[0] = (int16_t)bytestream_get_le16(&buf);
  198. }
  199. /* decode the samples */
  200. while (buf < buf_end) {
  201. predictor[ch] += s->roq_square_array[*buf++];
  202. predictor[ch] = av_clip_int16(predictor[ch]);
  203. *output_samples++ = predictor[ch];
  204. /* toggle channel */
  205. ch ^= stereo;
  206. }
  207. break;
  208. case CODEC_ID_INTERPLAY_DPCM:
  209. buf += 6; /* skip over the stream mask and stream length */
  210. for (ch = 0; ch < s->channels; ch++) {
  211. predictor[ch] = (int16_t)bytestream_get_le16(&buf);
  212. *output_samples++ = predictor[ch];
  213. }
  214. ch = 0;
  215. while (buf < buf_end) {
  216. predictor[ch] += interplay_delta_table[*buf++];
  217. predictor[ch] = av_clip_int16(predictor[ch]);
  218. *output_samples++ = predictor[ch];
  219. /* toggle channel */
  220. ch ^= stereo;
  221. }
  222. break;
  223. case CODEC_ID_XAN_DPCM:
  224. {
  225. int shift[2] = { 4, 4 };
  226. for (ch = 0; ch < s->channels; ch++)
  227. predictor[ch] = (int16_t)bytestream_get_le16(&buf);
  228. ch = 0;
  229. while (buf < buf_end) {
  230. uint8_t n = *buf++;
  231. int16_t diff = (n & 0xFC) << 8;
  232. if ((n & 0x03) == 3)
  233. shift[ch]++;
  234. else
  235. shift[ch] -= (2 * (n & 3));
  236. /* saturate the shifter to a lower limit of 0 */
  237. if (shift[ch] < 0)
  238. shift[ch] = 0;
  239. diff >>= shift[ch];
  240. predictor[ch] += diff;
  241. predictor[ch] = av_clip_int16(predictor[ch]);
  242. *output_samples++ = predictor[ch];
  243. /* toggle channel */
  244. ch ^= stereo;
  245. }
  246. break;
  247. }
  248. case CODEC_ID_SOL_DPCM:
  249. if (avctx->codec_tag != 3) {
  250. uint8_t *output_samples_u8 = data;
  251. while (buf < buf_end) {
  252. uint8_t n = *buf++;
  253. s->sample[0] += s->sol_table[n >> 4];
  254. s->sample[0] = av_clip_uint8(s->sample[0]);
  255. *output_samples_u8++ = s->sample[0];
  256. s->sample[stereo] += s->sol_table[n & 0x0F];
  257. s->sample[stereo] = av_clip_uint8(s->sample[stereo]);
  258. *output_samples_u8++ = s->sample[stereo];
  259. }
  260. } else {
  261. while (buf < buf_end) {
  262. uint8_t n = *buf++;
  263. if (n & 0x80) s->sample[ch] -= sol_table_16[n & 0x7F];
  264. else s->sample[ch] += sol_table_16[n & 0x7F];
  265. s->sample[ch] = av_clip_int16(s->sample[ch]);
  266. *output_samples++ = s->sample[ch];
  267. /* toggle channel */
  268. ch ^= stereo;
  269. }
  270. }
  271. break;
  272. }
  273. *data_size = out;
  274. return buf_size;
  275. }
  276. #define DPCM_DECODER(id_, name_, long_name_) \
  277. AVCodec ff_ ## name_ ## _decoder = { \
  278. .name = #name_, \
  279. .type = AVMEDIA_TYPE_AUDIO, \
  280. .id = id_, \
  281. .priv_data_size = sizeof(DPCMContext), \
  282. .init = dpcm_decode_init, \
  283. .decode = dpcm_decode_frame, \
  284. .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
  285. }
  286. DPCM_DECODER(CODEC_ID_INTERPLAY_DPCM, interplay_dpcm, "DPCM Interplay");
  287. DPCM_DECODER(CODEC_ID_ROQ_DPCM, roq_dpcm, "DPCM id RoQ");
  288. DPCM_DECODER(CODEC_ID_SOL_DPCM, sol_dpcm, "DPCM Sol");
  289. DPCM_DECODER(CODEC_ID_XAN_DPCM, xan_dpcm, "DPCM Xan");